Have a strange issue at a customer, they went and replaced all of their old PoE switches with brand new HPE 5130 EI Switch Series.Their PBX has been up and stable for several years with no recent changes, but since they change the switches they are hav..
I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages :$ sudo dpkg -l|grep -Ei dahdi|asterisk|libpriiiasterisk 1:11.13.1~dfsg-2+b1 amd64Open Source Private Branch Exchange (PBX)iiasterisk-config1:11.13.1~dfsg-2allConfigurat..
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually reach the PBX, but for some reason, they are not caught by any of my extensions context.He..
When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …Is there a way to force URI calls through the PBX? I have fo..
I have a somewhat confusing use case.We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine. Sometimes however, our users are also connected to our ..
friends:I am facing cutoffs randomly when negotiating calls.The PBX dials the destination, the provider (softswitch) receives the request ** and sudenly the PBX hangs up the call* * while the provider is still dialing it, as a consequence the rem..